<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML xmlns="http://www.w3.org/TR/REC-html40" xmlns:v =
"urn:schemas-microsoft-com:vml" xmlns:o =
"urn:schemas-microsoft-com:office:office" xmlns:w =
"urn:schemas-microsoft-com:office:word" xmlns:st1 =
"urn:schemas-microsoft-com:office:smarttags"><HEAD>
<META http-equiv=Content-Type content="text/html; charset=us-ascii">
<META content="MSHTML 6.00.2900.2722" name=GENERATOR><o:SmartTagType
name="metricconverter"
namespaceuri="urn:schemas-microsoft-com:office:smarttags"></o:SmartTagType><!--[if !mso]>
<STYLE>st1\:* {
BEHAVIOR: url(#default#ieooui)
}
</STYLE>
<![endif]-->
<STYLE>@font-face {
font-family: Wingdings;
}
@page Section1 {size: 595.3pt 841.9pt; margin: 3.0cm 2.0cm 3.0cm 2.0cm; }
P.MsoNormal {
FONT-SIZE: 12pt; MARGIN: 0cm 0cm 0pt; FONT-FAMILY: "Times New Roman"
}
LI.MsoNormal {
FONT-SIZE: 12pt; MARGIN: 0cm 0cm 0pt; FONT-FAMILY: "Times New Roman"
}
DIV.MsoNormal {
FONT-SIZE: 12pt; MARGIN: 0cm 0cm 0pt; FONT-FAMILY: "Times New Roman"
}
H2 {
FONT-SIZE: 14pt; MARGIN: 12pt 0cm 3pt; FONT-FAMILY: Arial
}
A:link {
COLOR: blue; TEXT-DECORATION: underline
}
SPAN.MsoHyperlink {
COLOR: blue; TEXT-DECORATION: underline
}
A:visited {
COLOR: purple; TEXT-DECORATION: underline
}
SPAN.MsoHyperlinkFollowed {
COLOR: purple; TEXT-DECORATION: underline
}
SPAN.EmailStyle17 {
COLOR: windowtext; FONT-FAMILY: Arial; mso-style-type: personal-compose
}
SPAN.Overskrift2Tegn {
FONT-WEIGHT: bold; FONT-FAMILY: Arial
}
DIV.Section1 {
page: Section1
}
OL {
MARGIN-BOTTOM: 0cm
}
UL {
MARGIN-BOTTOM: 0cm
}
</STYLE>
<!--[if gte mso 9]><xml>
<o:shapedefaults v:ext="edit" spidmax="1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext="edit">
<o:idmap v:ext="edit" data="1" />
</o:shapelayout></xml><![endif]--></HEAD>
<BODY lang=DA vLink=purple link=blue>
<DIV dir=ltr align=left><SPAN class=641480019-17102005><FONT face=Arial
color=#0000ff size=2>Okay so it seems like it was the firewall, someone just
suggested that we disable it (On Redhat server) and it's working fine... so does
anyone know clearly what ports (other than 5060) SIP uses for these
calls?</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=641480019-17102005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=641480019-17102005><FONT face=Arial
color=#0000ff size=2>-- Mike</FONT></SPAN></DIV><BR>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Michael
Furdyk<BR><B>Sent:</B> October 17, 2005 2:54 PM<BR><B>To:</B> Asterisk Users
Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> [Asterisk-Users] SIP
to SIP sadness<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV dir=ltr align=left><SPAN lang=EN-GB
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p><SPAN
class=224024903-17102005>Wow, after getting the O'Reilly book delivered last
week along with two Digium TDM400P's, I'm really getting the hang of this.
But the SIP to SIP issue is still a problem... and it seems silly because
everything else (should have been?) so much harder but is working pretty
flawlessly. Basically I get no audio either way, and it tries to do a "native
bridge" (handoff?)</SPAN></o:p></SPAN></DIV>
<DIV dir=ltr align=left><SPAN lang=EN-GB
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p><SPAN
class=224024903-17102005></SPAN></o:p></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN lang=EN-GB
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p><SPAN
class=224024903-17102005>So when I dial another SIP extension, I
get:</SPAN></o:p></SPAN></DIV>
<DIV dir=ltr align=left><SPAN lang=EN-GB
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p><SPAN
class=224024903-17102005></SPAN></o:p></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN lang=EN-GB
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p><SPAN
class=224024903-17102005></SPAN></o:p></SPAN> <FONT face=Arial
size=2>---<BR> -- SIP/324-ab4d answered SIP/322-7e8d<BR>We're
at 192.168.1.195 port 16874<BR>Answering with preferred capability 0x2
(gsm)<BR>Answering with preferred capability 0x4 (ulaw)<BR>Answering with
non-codec capability 0x1 (telephone-event)<BR>Reliably Transmitting (NAT) to
192.168.1.24:5060:<BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP
192.168.1.24:5060;branch=z9hG4bK8E845F95F34044ACA77C54EF28288C32;received=192.168.1.24;rport=5060<BR>From:
Michael Furdyk <sip:322@192.168.1.195>;tag=411158625<BR>To:
<sip:324@192.168.1.195>;tag=as6606adb1<BR>Call-ID: <A
href="mailto:28E78AC1-5FDE-414E-8059-68B393A24F60@192.168.1.24">28E78AC1-5FDE-414E-8059-68B393A24F60@192.168.1.24</A><BR>CSeq:
30931 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER, NOTIFY<BR>Contact: <sip:324@192.168.1.195><BR>Content-Type:
application/sdp<BR>Content-Length: 239</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial size=2>v=0<BR>o=root 3348 3348 IN IP4
192.168.1.195<BR>s=session<BR>c=IN IP4 192.168.1.195<BR>t=0 0<BR>m=audio 16874
RTP/AVP 3 0 101<BR>a=rtpmap:3 GSM/8000<BR>a=rtpmap:0 PCMU/8000<BR>a=rtpmap:101
telephone-event/8000<BR>a=fmtp:101 0-16<BR>a=silenceSupp:off - - -
-</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial size=2>---<BR> --
Attempting native bridge of SIP/322-7e8d and SIP/324-ab4d</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial size=2><-- SIP read from
192.168.1.24:5060: <BR>ACK sip:324@192.168.1.195 SIP/2.0<BR>Via: SIP/2.0/UDP
192.168.1.24:5060;rport;branch=z9hG4bKB644BADE71EF4422878597A96BE8D613<BR>From:
Michael Furdyk <sip:322@192.168.1.195>;tag=411158625<BR>To:
<sip:324@192.168.1.195>;tag=as6606adb1<BR>Contact:
<sip:322@192.168.1.24:5060><BR>Call-ID: <A
href="mailto:28E78AC1-5FDE-414E-8059-68B393A24F60@192.168.1.24">28E78AC1-5FDE-414E-8059-68B393A24F60@192.168.1.24</A><BR>CSeq:
30931 ACK<BR>Max-Forwards: 70<BR>Content-Length: 0</FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial size=2></FONT> </DIV>
<DIV dir=ltr align=left><SPAN class=224024903-17102005><FONT face=Arial
size=2>Here is my default in SIP.conf. Each SIP config has
canreinvite=no</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=224024903-17102005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=224024903-17102005><FONT face=Arial
size=2>[general]<BR>disallow=all<BR>allow=gsm<BR>allow=ulaw<BR>nat=no<BR>canreinvite=no<BR>externip=(real
external IP is
here)<BR>localnet=192.168.1.195/255.255.255.0<BR>srvlookup=yes<BR>sipdebug=yes<BR></FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=224024903-17102005><FONT face=Arial size=2>I
have tried nat=no and nat=yes</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=224024903-17102005><FONT face=Arial
size=2> </DIV></FONT></SPAN></BODY></HTML>